JP2000078018A - Voice coding system and device and data recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To produce the coded data of high sound quality in real time and with no sound break by switching the using bit allocation means to execute the bit allocation and then the coding so as to perform the due processing by means of the prescribed one of plural bit allocation means. SOLUTION: A sub-band analyzing means 102 divides the inputted digital audio signals into 32 frequency components. A grouping means 111 divides the 32 frequency components into the number of groups that is designated by the throughput control information 121. An allocatable bit arithmetic means 113 decides each allocatable bit number to an aural psychological model bit allocation means 110 and a band output adaptive bit allocation means 109 based on the total allocatable bit number. Each of sub-band signals is coded by a quantizing/coding means 106 and an auxiliary information coding means 107, and a bit stream forming means 108 forms and outputs a bit stream.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

[0001] The present invention relates to a speech coding system,
And an audio encoding device and a data storage medium, particularly an audio encoding method using a subband encoding method as used in the MPEG (Motion Picture Experts Group) method, an audio encoding device, and the audio code The present invention relates to a data storage medium storing a program for executing a conversion method.

[0002]

2. Description of the Related Art In recent years, with the use of multimedia in personal computers and the spread of the Internet, an environment in which moving images and audio such as MPEG can be reproduced by software on personal computers (hereinafter, also referred to as PCs) has been prepared. MPEG
And the like, the use range of encoded data is expanding. However, with respect to encoders for producing encoded data, expensive hardware is still mainly used.
In some cases, the encoded data is created by software, but the encoding takes a lot of processing time compared to the real time of the video or audio to be encoded, and the encoding takes a lot of time. , And it is troublesome and has not yet spread widely.

[0003] For this reason, there is a demand for realizing encoded data in real time by software processing in order for ordinary personal computer users to be able to create encoded data easily and cheaply.

[0004] An example of a conventional speech coding system will be described below. FIG. 11 shows an M format standardized by ISO / IEC11172-3 as an encoded data format related to audio.
FIG. 2 is a block diagram of a PEG audio encoder. FIG.
In 1, the input digital audio signal is divided into 32 frequency components by a sub-band analysis unit 202, and a scale factor is calculated for each sub-band signal by a scale factor extraction unit 203 to make the dynamic range uniform. The input digital audio signal is subjected to a fast Fourier transform by the FFT unit 204.
(FFT: Fast Fourier Transform), and using the result, the signal-to-mask ratio (SM) based on the psychoacoustic model utilizing the characteristics of human hearing by the psychoacoustic analysis means 205
R) Using the value relation model, the bit allocation means 20
The number of bits allocated to each sub-band signal is determined by 6. Quantization / encoding means 207 quantizes / encodes each subband signal in accordance with the number of bits allocated to each subband signal. Then, the bit stream forming unit 209 forms and outputs a bit stream by using the header information and the auxiliary information encoded by the auxiliary information encoding unit 208 together.

[0005] Since this conventional speech coding system is a coding system in which coding is performed for each band (sub-band) using uneven distribution of power in each band, each sub-band using an psychoacoustic model is used. The bit allocation to the band signal affects the sound quality. In addition, since it was standardized for the purpose of using storage media, it is suitable for high-quality sound of coded data, but not suitable for real-time coding. The calculation amount is large.

[0006]

The conventional speech coding system and the conventional speech coding apparatus are constructed as described above, and are suitable for producing high-quality coded data for a storage medium. However, since the use of the psychoacoustic model requires a large amount of processing capability, the current CPU capability for processing by software is inappropriate for real-time processing on a PC. Also, a high-performance CPU capable of real-time processing
Even if the application is operated on a PC equipped with the application, when the CPU occupancy of another application becomes large, there is a possibility that the processing in real time may become impossible, and as a result, the sound is interrupted. There is a problem that there is a possibility.

The present invention has been made in order to solve the above problems, and has a CPU processing capability of a personal computer.
And a voice coding method and a voice coding device capable of realizing, by software processing, creation of high-quality coded data in which sound is not interrupted in real time without being affected by the CPU occupancy of another application. An object of the present invention is to provide a data storage medium storing a program for executing encoding.

[0008]

According to a first aspect of the present invention, there is provided a voice coding system for dividing a digital audio signal into a plurality of frequency bands and performing coding for each band. Generating bit allocation information for each of the divided bands, having a plurality of bit allocation units each having a different processing amount, and determining a predetermined bit from among the plurality of bit allocation units based on external control information. In order to perform processing using the allocating means, the bit allocating means to be used is switched to execute bit allocation and perform encoding.

According to a second aspect of the present invention, there is provided a speech encoding system according to the first aspect, wherein the central processing unit can occupy for performing an encoding process as the external control information. Using the load value representing the processing amount of the processing device, based on the load value, the processing amount when performing the encoding operation using each bit allocating unit in the encoding process in the central processing unit is stored in advance. With reference to the data table, the bit allocation means is selected so that the processing amount of the central processing unit that can be occupied by the encoding processing does not exceed the processing amount.

According to a third aspect of the present invention, there is provided the voice encoding system according to the second aspect, wherein the load value is a central processing process which can be occupied for performing a coding process. It uses processing amount control information from monitoring means for monitoring the processing amount of the apparatus.

According to a fourth aspect of the present invention, in the audio coding system according to the first aspect of the present invention, high quality sound of the encoded data is realized as the bit allocation processing by the bit allocation means. A process using a high-efficiency bit allocation method for allocating bits at a possible high efficiency, and a low-load bit allocation method for performing bit allocation with a low load with a small processing amount as compared with the process using the above-mentioned high efficiency bit allocation method Is performed.

According to a fifth aspect of the present invention, there is provided the audio coding system according to the first aspect, wherein the switching of the bit allocation means used at the time of the encoding can be decoded into an audio signal. This is performed on a frame basis which is a minimum unit.

According to a sixth aspect of the present invention, there is provided the speech encoding system according to the first aspect, wherein the sub-band signals of each band divided into a plurality of frequency bands are respectively predetermined. The group is divided into groups each having a predetermined number of sub-band signals, and bit allocation processing is performed independently for each group to generate bit allocation information for each band.

According to a seventh aspect of the present invention, in the audio coding system according to the sixth aspect, the grouping is performed in the number of groups or in the frequency axis direction within the groups. The number of subband signals is variably set to the number specified by the external control information or the number specified based on the processing amount control information from the monitoring unit.

According to an eighth aspect of the present invention, in the audio coding system according to the seventh aspect, the process of changing the number of the sub-band signals is a minimum unit which can be decoded into an audio signal. Is performed in frame units.

According to a ninth aspect of the present invention, there is provided the audio coding system according to the eighth aspect, wherein at least one group to which no bit is allocated is provided at the time of the grouping. .

According to a tenth aspect of the present invention, in the audio coding system according to the sixth aspect, the subband signals are grouped into groups belonging to a low band by grouping the subband signals. For a subband signal, using a highly efficient bit allocation method capable of realizing high quality sound of encoded data, assigning bits to subband signals within a group and assigning bits to subband signals within a group For the sub-band signals grouped into the group belonging to the high band, the bits are allocated to the sub-band signals in the group by using a low-load bit allocation process with a smaller processing amount compared to the high-efficiency bit allocation method Things. An audio encoding method according to claim 11 of the present invention is the audio encoding method according to claim 6, wherein the assignable bit calculating means determines the number of assignable bits for the independent bit allocating means for each group. And using the ratio of each group to the entire group, taking into account the weight based on the characteristics of each band belonging to each group, and assigning the number of bits that can be allocated to the entire group to the bit allocation means independent for each group. It is something to sort.

According to a twelfth aspect of the present invention, in the audio coding system according to the eleventh aspect, weighting based on characteristics of each band belonging to each group is performed for each band. The weighting is based on a predetermined minimum audible limit value. Claim 13 of the present invention
The audio coding method according to claim 11, wherein each group obtained by performing a sub-band analysis on an input digital audio signal is weighted based on a characteristic of each band belonging to each group. Are weighted based on the subband signal level of each frequency band belonging to.

According to a fourteenth aspect of the present invention, in the audio coding system according to the eleventh aspect, weighting based on characteristics of each band belonging to each group is performed by the input digital audio signal. Are weighted based on the spectral signal level belonging to each group obtained by performing a linear transformation on

According to a fifteenth aspect of the present invention, in the voice coding system according to the sixth aspect of the present invention, a high level signal whose signal level belonging to each group is equal to or higher than a predetermined threshold value is obtained. On the other hand, bit allocation is performed using a high-efficiency bit allocation method that can realize high sound quality of encoded data, and the signal level belonging to each group is
For low-level signals equal to or lower than a predetermined threshold value, bit allocation is performed using a low-load bit allocation method that requires less processing than the above-described high-efficiency bit allocation method.

According to a sixteenth aspect of the present invention, in the audio coding system according to the fifteenth aspect, a signal level belonging to each of the groups is subjected to sub-band analysis of an input digital audio signal. The sub-band signal level is obtained as follows.

According to a seventeenth aspect of the present invention, in the audio coding system according to the fifteenth aspect, the signal level belonging to each group is obtained by performing a linear conversion on an input digital audio signal. This is the spectral signal level obtained.

[0023] In the speech coding system according to claim 18 of the present invention, in the speech coding system according to claim 15, the signal level belonging to each group is set to a minimum audible threshold value for each predetermined band. It is assumed that.

According to a nineteenth aspect of the present invention, in the speech coding system according to any one of the fourth, tenth, and fifteenth aspects, it is possible to realize high-quality sound of the encoded data. The high-efficiency bit allocation process is a bit allocation process that is performed using a relationship between a signal and a mask ratio value based on a predetermined psychoacoustic model. The load bit allocation process is a bit allocation process performed by adding a predetermined minimum audible limit value for each band to a signal level divided into a plurality of frequency bands.

[0025] According to a twentieth aspect of the present invention, in the above-mentioned nineteenth aspect, the psychoacoustic model is MPEG (Motion Picture Exposure).
ertsGroup).

According to a twenty-first aspect of the present invention, in the audio coding system according to the fifth or eighth aspect, a frame which is a minimum unit that can be decoded into the audio signal is an MPEG signal. (Motion Picture Exp
erts Group).

[0027] According to a speech encoding system of the present invention, in the speech encoding system of the first aspect, the bit allocating means may include a predetermined auditory psychology for each of the divided bands. The bit allocation information is generated based on the information output from the model, and N (N =
Once every 1,2,3 ...) frame, bit allocation information is generated based on information output from the predetermined psychoacoustic model. For a frame for which the bit allocation information is not generated, Bit allocation information is generated based on the information output from the psychoacoustic model and the signal information of each of the divided bands, and is encoded.

According to a twenty-third aspect of the present invention, there is provided a speech encoding system according to the first aspect, wherein the speech encoding system has an auditory psychological model capable of controlling a processing amount in a stepwise manner. Based on the control information, the processing amount of the psychoacoustic model is controlled, and bit allocation information for each band is generated so that the processing is performed using the psychoacoustic model having a predetermined processing amount.

A speech coding method according to a twenty-fourth aspect of the present invention is the speech coding method according to the first aspect, wherein a plurality of psychoacoustic models having different processing amounts are provided, and control information from outside is provided. On the basis of the plurality of psychoacoustic models, a psychoacoustic model to be used is switched so that processing is performed using a predetermined psychological model, and bit allocation information for each band is generated. .

Further, according to the speech coding system of the present invention, a digital audio signal is divided into a plurality of frequency bands, bit allocation information for each of the divided bands is generated, A voice coding method that performs coding for each band for transmission purposes, and controls the bit allocation range in a frame in which data is inserted into a coded data stream, and variably controls the amount of coded audio data. It is.

According to a speech coding method according to claim 26 of the present invention, in the speech coding method according to claim 25, a bit allocation range is controlled in frame units based on external control information. It controls the amount of encoded audio data variably.

According to a twenty-seventh aspect of the present invention, in the audio coding system according to the twenty-sixth aspect, as the external control information, a monitor means for monitoring a buffer of additional data is provided. It uses data amount control information.

According to a twenty-eighth aspect of the present invention, in the voice coding system according to the first aspect, the coding is performed in accordance with the performance of the central processing unit in which the coding process is performed. At the time of initialization before the processing operation, the processing load value information of a plurality of bit allocation means or a plurality of psychoacoustic models is output to the outside.

According to a twenty-seventh aspect of the present invention, in the speech coding system according to the twenty-eighth aspect, a plurality of bit allocation means output as information to the outside, or a plurality of psychoacoustic systems are provided. Each processing load value information of the model is output in descending order or ascending order.

The audio coding method according to claim 30 of the present invention is different from the audio coding method in which the video signal and the audio signal are encoded by the same central processing unit. The encoding is performed by the amount of operation, and the overall amount of operation of the processing on the central processing unit is controlled by changing the amount of operation of encoding the audio signal or the video signal. is there.

The audio coding method according to claim 31 of the present invention is the same as the audio coding method in which the video signal and the audio signal are processed by the same central processing unit. The encoding is performed using a plurality of different encoding methods, and by changing the encoding method of the audio signal encoding, the overall operation amount of the processing on the central processing unit is reduced. To control.

According to a thirty-second aspect of the present invention, in the audio coding system according to the thirty-first or thirty-first aspect, the control of the processing on the central processing unit is controlled by an external control. This is performed based on information.

[0038] Further, a voice coding method according to claim 33 of the present invention is a voice coding method for performing time / frequency conversion on a digital audio signal and generating quantization information to perform coding. There is provided a plurality of quantization information calculation means each having a different operation amount, and a predetermined quantization information calculation means is selected from the plurality of quantization information calculation means based on external control information. In order to perform the processing, the quantization information calculating means to be used is switched to calculate the quantization information and perform the encoding.

A speech coding apparatus according to a thirty-fourth aspect of the present invention performs speech coding using the speech coding method according to any one of the first to thirty-third aspects.

A recording medium according to a thirty-fifth aspect of the present invention has recorded thereon the steps of the audio encoding system according to any one of the first to thirty-third aspects.

[0041]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS A speech coding system and a speech coding apparatus according to an embodiment of the present invention will be described below with reference to the drawings. (Embodiment 1) Here, for example, in the case of an encoding system in which an input signal is divided into a plurality of frequency components, and encoding is performed for each band (subband) using uneven distribution of power in each band. Will be described. FIG. 1 is a conceptual diagram of the entire system in the case where a personal computer (hereinafter, also referred to as a PC) is used as an audio encoding device according to the above encoding method. A hard disk drive (HDD) 11, which is a so-called multimedia type PC capable of inputting data from an external device, has a large storage capacity for storing various data and programs, and is a fixed recording medium;
PD which is a detachable storage medium having a relatively small storage capacity for inputting / outputting programs, data, etc. from D11
Drive 12a, FD drive 12b, and the HD
D is stored in a central processing unit (CP)
U) It is configured to be appropriately read out and executed on a memory 13 constituted by a random access memory (RAM) or the like by an instruction from the U) 14. In addition, a video capture card 16 and a sound card 18 are respectively incorporated in the camera 17 and the microphone 19 as external devices in order to capture images and sounds. The components of the PC 1 having the above configuration are connected by the internal data bus 15.

FIG. 2 is a block diagram of the encoder 20 of the audio encoding apparatus for implementing the audio encoding process executed by the PC 1 shown in FIG.
This is realized by a program read from the memory 13 into the memory 13. In FIG. 2, reference numeral 21 denotes a CPU1.
Reference numeral 4 denotes CPU load monitoring information for monitoring the load state. Reference numeral 22 denotes an encoding unit that controls the operations of the low-band coding processing unit 23 and the high-band coding processing unit 24 based on the CPU load monitoring information 21. Control means. Reference numeral 25 denotes a bit stream forming processing unit for converting the outputs of the two encoding processing units 23 and 24 into stream signals. Further, reference numeral 26 denotes an encoding mode designation signal input to the encoding means control means 22 when designated by the user.

As the configuration of the low-band coding processing means 23 in FIG. 2, for example, the configuration as shown in the conventional example of FIG. 11 is used. Also, as shown in FIG. 3, for example, as shown in FIG. 3, the configuration of the high-band coding processing unit 24 performs coding for each band (sub-band) using the uneven distribution of each band power. However, the bit allocation to each sub-band signal using the psychoacoustic model is not performed, and a band output adaptive bit allocating unit 304 is provided as an alternative to the sub-band signal. The scale factor is weighted based on human auditory characteristics, the first purpose is to perform low-load processing rather than high sound quality, and the configuration is such that the amount of calculation is small. Also, in order to eliminate excessive concentration of bit allocation to a specific band, weight adjustment according to each band is performed for each bit allocation.

FIG. 4 is a block diagram showing a detailed configuration of the encoder 20 shown in FIG. 2. Reference numeral 101 denotes an encoder, which is a subband analyzing means 102, a scale factor extracting means 103, and an FFT means, which will be described later. 104, psychoacoustic analysis means 105, quantization / encoding means 106, auxiliary information encoding means 107, bit stream forming means 108, band output adaptive bit allocating means 109, psychoacoustic model bit allocating means 110, grouping means 111, It comprises a bit allocation processing control means 112 and an allocatable bit operation means 113.

The sub-band analyzer 102 divides the input digital audio signal into 32 frequency components. The scale factor extracting means 103 calculates a scale factor for each sub-band signal and aligns each sub-band dynamic range. The grouping unit 111 divides the 32 divided frequency components into the number of groups specified by the processing amount control information 121 which is external control information. In the first embodiment, as shown in FIG. 5, the number of groups is 3, and each group is a low-band group A of 0 to 15 sub-bands and 16 to 29 sub-bands as sub-band signals continuous in the frequency axis direction. High band group B, and 30 to not perform bit allocation
Grouping into invalid group C of 31 subbands. It is assumed that the processing amount control information 121 includes information of the CPU load monitoring information 21 and the encoding mode setting signal 26. Further, in the first embodiment, as a bit allocating means for allocating bits to each of the sub-band groups, a low band that is sensitive to the human ear is
Psychoacoustic model bit allocation means 110 for highly efficient bit allocation using the relationship between signal and mask ratio values based on the psychoacoustic model specified by MPEG
Using a high-frequency band relatively low in sensitivity to the human ear, and adding a preset minimum audible limit value for each band to the scale factor information from the scale factor extracting means 103. The configuration is such that band output adaptive bit allocating means 109 for allocating bits with a lower load as compared with the psychoacoustic model bit allocation method is used.

The bit allocation processing control means 112
Is to perform an FFT (Fast Fourier Transform) means on the input digital audio signal to perform a required psychoacoustic analysis on a low-band group A of 0 to 15 sub-bands for which psychoacoustic model bits are allocated. At 104, the FFT means 104 is controlled so as to perform fast Fourier transform. Then, using this conversion result, psychoacoustic analysis means 1
05 derives a relational model of a signal-to-mask ratio (SMR) value based on a psychoacoustic model utilizing characteristics of human hearing.

The assignable bit operation means 113
Calculates the number of bits that can be allocated to the entire group determined from the sampling frequency and the bit rate value of the coding, based on the characteristics of each band belonging to each group, based on the ratio of each group to be allocated to the entire group. The number of bits that can be allocated to the independent bit allocation means is calculated for each group by using the weighted values. In the first embodiment, the whole psychological model is assigned to the psychoacoustic model bit allocating unit 110 and the band output adaptive bit allocating unit 113 in consideration of the scale factor index value and the ratio of the low band / high band region. From the possible number of bits, both means 11
The number of bits that can be allocated to each of 0 and 113 is determined. That is, actually, each scale factor index value obtained by the scale factor extraction means 103
From scf_index [i], as shown in the following Expressions 1 and 2, the added value Vpsy, V of scf_index [i] in both regions is obtained.
Calculate non.

(Equation 1)

(Equation 2) Here, psy_end = 16: the number of subbands for which the psychoacoustic model bits are allocated subband_end = 30: the number of subbands to which all bits are allocated.

Next, to a low band that is sensitive to the human ear,
In order to allocate more bits, Vpsy is weighted, and Vpsy = Vpsy * 0.75 perceptual psychological model bit assignable number psy_num and band output adaptive bit assignable number non_num are obtained based on the following equations. Vnon = Vnon * psy-ratio psy_num = all_alloc_num * Vnon / (Vpsy + Vnon) non_num = all_alloc_num-psy_num where all_alloc_num: total number of allocatable bits psy_ratio: psy_end / ( subband_end-psy_end).

The psychoacoustic model bit allocating means 110 receives the SM from the psychoacoustic analyzing means 105 within the range of the number of assignable bits (psy_num, non_num) for each group.
Using the relation model of the R value, the low band group A of 0
Bit allocation is performed for ~ 15 subbands. On the other hand, the band output adaptive bit allocating unit 109 performs bit allocation on the 16 to 29 subbands that are the high band group B. In addition, no bits are assigned to the 30 to 31 subbands that are invalid group C because they are invalid subbands.

According to the number of bits allocated to each subband signal determined by these bit allocation means,
Each sub-band signal is quantized / encoded by the quantization / encoding means 106, and the header information and the auxiliary information encoded by the auxiliary information encoding means 107 are formed into a bit stream by the bit stream forming means 108. Output.

When the information from the processing amount control information 121 is, for example, information for reducing the encoding processing amount, as shown in FIG. The bandwidth of the 0-15 sub-band low band group A to be allocated to the bits is
0-7 sub-band low band group A ', and conversely, a band output adaptive bit allocating means 10 with a small processing amount.
The bandwidth subject to the bit allocation of 9 is 8 to 29.
It is increased so as to be the sub-band low band group B ′. Furthermore, when the encoding processing amount is considered to decrease, the processing amount is controlled by setting the bit allocation target of the band output adaptive bit allocation means 109 to 0 to 29 subband groups. In this case, since the psychoacoustic model bit assignment means 110 does not substantially operate, the FFT means 104 and the psychoacoustic analysis means 10
5 will also not work.

On the other hand, if the information from the processing amount control information 121 is, for example, information for improving the sound quality of the coded data, the psychoacoustic which enables highly efficient (high sound quality) bit allocation. The bandwidth to which the model bit allocation means 110 performs bit allocation is increased. further,
The final form when considering high sound quality is that the bit allocation target of the psychoacoustic model bit allocation means 110 is 0 to 0.
29 subband groups. In this embodiment, as described above, the number of sub-band groups is increased or decreased and the bit allocation means is switched in units of frames, which is the minimum unit that can be decoded into an audio signal, so that the amount of encoding processing can be controlled in real time. .

Next, an overall operation flow of the speech coding apparatus according to the first embodiment will be described with reference to FIG. First, using the configuration shown in FIG. 7A, in order to recognize the processing load of each encoder (23, 24), in each mode of each encoder (change in the bandwidth to be allocated to bits) By encoding dummy data for a predetermined time, the CPU load monitoring unit 700
Stores the CPU load value in each mode in the data table 701. When the sample (data) is input, in step S70 of FIG. 7B, sub-band analysis is performed to divide the frequency components into 32 frequency components. Then, in step S71, the scale factor of each sub-band signal is Is calculated.

Next, in step S72, the CPU
It is determined whether or not there is load detection data. Here, since the operation has just started, there is no CPU load detection data. Therefore, the process proceeds to step S74, where normal grouping that can perform the highest-quality sound reproduction is performed. Then, the process proceeds to step S75, where the psychoacoustic model bit allocation process is performed. Then, the process proceeds to step S76 to perform quantization /
An encoding process is performed, and a bit stream is formed in step S79 to complete a series of processes. At the end of the process, a time required for encoding a predetermined number of input samples is determined by the CPU load monitoring unit 700. And the current CPU load is detected.

Then, in the processing from the next time, the CPU load detection is determined to be "present" in step S72, and in step S73, the detected CPU load cannot be encoded in real time. If it is determined that the mode is not the same, the process proceeds to step S77, where the optimum mode (grouping) is selected with reference to the data table 701, and the psychological psychology in step S75 is performed together with the band bit allocation process in step S78. Each of the model bit allocation processes is performed at a predetermined ratio, and the process proceeds to step S76 where quantization / encoding process is performed. In step S79, a bit stream using these encoded data is formed. Become.

The operation of the assignable bit operation means 113 according to the first embodiment is performed by considering the scale factor index value and the ratio of the low band and the high band to the independent bit allocation for each group. Although the number of bits that can be assigned to the means is calculated, the level of the spectrum signal belonging to each group from the FFT means 104 may be used instead of the scale factor index value. It may be.

The encoder 10 according to the first embodiment
1 is provided inside the encoder 101 with CPU load monitoring means 700 for monitoring the processing amount of the CPU, and operates the encoder 101 so as not to exceed the processing capacity of the CPU. Although it is configured to perform the control, external control information such as a user input may be used. By performing the user input, it is possible to perform the encoding process in which the sound quality and the image quality are prioritized according to the user's preference.

The encoder 10 according to the first embodiment
As one bit allocation unit, an psychoacoustic model bit allocation unit 110 that performs bit allocation with high efficiency in a low band that is sensitive to the human ear is used.
Although the band output adaptive bit allocating means 109 for performing bit allocation with low load is fixedly used, the signal from the scale factor extracting means 103 causes the sub-band signal level belonging to each group to be set to a predetermined level for each band. In the case of being equal to or less than the threshold value, that is, as shown in FIG. 8, when the number of signals that are significant as encoded data in the low band is smaller than that in the high band, the bit allocating means is fixed according to the band. It is not necessary to use the psychoacoustic model bit allocation means 110 for a high band.

Also, as shown in FIG. 9, the signal from the scale factor extracting means 103 does not compare and determine the subband signal level belonging to each group with the threshold value. Also, the signal from the FFT means 104 having a high resolution (of high frequency) is input to the bit allocation processing control means 112,
The configuration may be such that a comparison between a level of a subband signal belonging to each group and a preset threshold value of each band is performed.

(Embodiment 2) Next, a data recording medium according to Embodiment 2 of the present invention will be described. By recording an encoding program for realizing the configuration of the audio encoding device or the encoding method shown in the first embodiment on a data storage medium such as a floppy disk, each embodiment of the present invention is realized. The processing described in the embodiment can be easily performed by an independent computer system.

That is, FIG. 10 is a diagram for explaining a case where the encoding process of the first embodiment is executed by a computer system using a floppy disk storing the encoding program. FIG. 10 (a)
FIG. 10B shows the appearance, cross-sectional structure, and floppy disk main body viewed from the front of the floppy disk, and FIG.
Shows an example of the physical format of the floppy disk body.

The above-mentioned floppy disk FD is composed of the above-mentioned floppy disk main body D and a floppy disk case FC.
A plurality of tracks Tr are formed concentrically from the outer circumference toward the inner circumference on the surface of the floppy disk main body D, and each track Tr has 16 sectors Se in the angular direction. Is divided into Therefore,
In the floppy disk FD storing the program, the floppy disk main body D has data as the program recorded in an area (sector) Se allocated thereon.

FIG. 10C shows a configuration for recording the program on the floppy disk FD and performing audio encoding processing using the program stored on the floppy disk FD. When the above program is recorded on the floppy disk FD, data as the above program is transmitted from the computer system Cs.
Writing to the floppy disk FD via the floppy disk drive FDD. In addition, floppy disk FD
When the speech encoding device is constructed in the computer system Cs by the program recorded in the computer system Cs, the program is read from the floppy disk FD by the floppy disk drive FDD, and the computer system Cs is read.
Load to s.

In the above description, the description has been made using a floppy disk as a data recording medium. However, even when an optical disk is used, audio encoding processing by software can be performed in the same manner as in the case of the floppy disk. The recording medium is not limited to the above-mentioned optical disk or floppy disk, but may be any one capable of recording a program such as an IC card, a ROM cassette, or the like. Similarly, voice coding processing by software can be performed.

(Embodiment 3) Next, a speech coding system and a speech coding apparatus according to Embodiment 3 of the present invention will be described with reference to the drawings. As the configuration of the high-band coding processing unit 24 shown in FIG. 2, for example, the configuration shown in FIG. 3 is used. Further, as shown in FIG. 12, for example, as shown in FIG. 12, the low-band coding processing means 23 uses the uneven distribution of each band power to code each band (sub-band). Although a coding method that performs encoding is used, bit allocation is not performed for each subband signal using only a predetermined psychoacoustic model analysis means, and a simplified psychoacoustic model unit 4062 with a small processing amount is newly provided.
Bit allocation can be performed from the masking threshold of the psychoacoustic model unit 4061 output in the previous frame and the bit allocation information generated based on the band division signal of the frame.

That is, FIG. 12 is a block diagram showing a detailed configuration of the low-band coding processing means 23 shown in FIG. 2. Reference numeral 401 denotes an encoder, which is a sub-band analyzing means 402 and a scale factor extracting means which will be described later. 403, bit scraping processing control means 404, FFT processing means 405,
It comprises a psychoacoustic analysis stage 406, a psychoacoustic model pit assigning means 407, a quantization / encoding means 408, an auxiliary information encoding means 409, and a bit stream forming means 410.

The operation will be described below. The sub-band analyzer 402 divides the input digital audio signal into 32 frequency components. The scale factor extracting means 403 calculates a scale factor for each sub-band signal and aligns each sub-band dynamic range. The FFT processing means 405 performs a fast Fourier transform on the input digital audio signal. The psychoacoustic analysis means 406 comprises, for example, a normal psychoacoustic model unit 4061 specified by MPEG and the simple psychoacoustic model unit 4062 having a smaller processing amount than the normal psychoacoustic model unit 4061. Calculate the mask ratio.

The normal auditory psychological model section 4061
Calculates the signal-to-mask ratio of each subband signal based on the following equation 3, whereas the simplified psychoacoustic model unit 4062 calculates the signal-to-mask ratio of each sub-hand in the corresponding frame as shown in the following equation 4. The minimum masking level is not calculated, and the minimum masking level in the latest previous frame calculated by the normal psychoacoustic model unit 4061 is used, and the sound pressure is determined by the scale extracted by the scale factor extracting unit 403 of the corresponding frame. Calculate the signal-to-mask ratio using the factor values.

(Equation 3) Here, Lsb (n): Ogo圧LT _min each subband (n): is the minimum masking level of each subband.

(Equation 4) Here, Lsb (n) = 20 · log (scf _max (n) · 32768-10) db scf _max (n): scale factor value for each subband of the corresponding frame LT _min (n): recent normal Auditory psychological model 4061
Minimum masking level of each subband calculated in

In the third embodiment, the bit allocation processing control means 404 sets N to 3 based on the information of the processing amount control information 121 as shown in FIG. The normal psychoacoustic model unit 4061 capable of outputting the optimal bit allocation information capable of realizing high quality with the psychoacoustic model unit 4062,
It controls how many frames are performed once and whether the FFT processing means 405 performs fast Fourier transform. For example, in the state of FIG. 13, when information indicating that the CPU occupation rate to be allocated to the encoding process is reduced is notified to the bit allocation processing control unit 404 as the information of the processing amount control information 121, To increase the use of the psychoacoustic model unit 4062, the value of N is increased. Conversely, when information indicating that the CPU occupation rate used for the encoding process may be used more is notified to the bit allocation processing control unit 404, the normal auditory psychological model unit 4061 capable of realizing high sound quality can be realized. Decrease the value of N for more use. Thereby, it is possible to control the processing amount.

Acoustic psychological model bit allocation means 407
Performs bit allocation to each sub-band signal divided by the sub-band analysis unit 402 based on the relationship between the signal and the mask ratio, which is information from the bit allocation processing control unit 404. Quantization / encoding means 408 performs quantization and encoding of each subband signal,
A bit stream is formed by the bit stream forming means 410 together with the auxiliary data from the auxiliary information encoding means 409, and is output.

As described above, according to the third embodiment, the bit allocation is performed once every N frames, so that the CPU load in the time axis direction can be reduced.

Although the encoder 401 is supposed to be the low-band coding processing means 23 shown in FIG. 2, it is not applied only to low-band signals but to all-band signals. You may make it apply.

(Embodiment 4) Next, a speech coding method and a speech coding apparatus according to Embodiment 4 of the present invention will be described with reference to the drawings. FIG. 14 uses an encoding method in which encoding is performed for each band (sub-band) using the uneven distribution of power in each band, as shown in FIG. The difference is that a function for adding external data other than audio data is provided. As the external data, image data and text data are assumed.

That is, the encoder 501 shown in FIG.
Are sub-band analysis means 502, scale factor extraction means 503, FFT processing means 504, psychoacoustic analysis means 505, bit allocation means 506, and quantization /
It comprises an encoding unit 507, an auxiliary information encoding unit 508, a bit stream forming unit 509, a bit allocation control unit 510, and an additional data encoding unit 511.

Hereinafter, the operation will be described. The sub-band analyzer 502 divides the input digital audio signal into 32 frequency components. The scale factor extracting means 503 calculates a scale factor for each subband signal, and aligns each subband dynamic range. The FFT processing means 504 performs a fast Fourier transform on the input digital audio signal. The psychoacoustic analysis unit 505 calculates a signal-to-mask ratio using a psychoacoustic model specified by MPEG, for example.

Bit allocation processing control means 510 monitors additional data buffer 512 for temporarily storing data to be added in the output bit stream, and determines whether or not there is additional data, or Based on the allocation range control information 513 generated from the determination as to whether or not to overflow, the bit allocation unit 506
Specify the range for bit allocation.

For example, when there is no data in the additional data buffer 512, as shown in FIG. 15, bits are allocated to the subbands 0 to 29. In this case, the total number of assignable bits is 100, and 80 bits are allocated to subbands 0 to 15 and 20 bits are allocated to subbands 16 to 29.

When data is written from the outside to the additional data buffer 512 and the additional data is present, that is, the allocation range control information 513
As a result, an instruction to insert additional data is notified to the bit allocation processing control means 510. In the fourth embodiment, for example, 80 bits are allocated to subbands 0 to 15, and subband 16 to be originally allocated is allocated. ~
No bit allocation is performed for 29 and the remaining 20
Bits are allocated as additional bits of data.
Also, for the sub-bands after the sub-band 16 for which bit allocation is not performed, the FFT processing and the psychoacoustic analysis of the corresponding range may not be performed in order to reduce the processing amount.

Then, quantization and encoding are performed on the sub-bands to which the above-mentioned bit allocation has been performed by the quantization / encoding unit 507, and the auxiliary data from the auxiliary information encoding unit 508 and, for example, MPEG A bit stream is formed and output by the bit stream forming means 509 together with the additional data coded as the ancillary data of.

As described above, according to the fourth embodiment, when performing transmission at a constant bit rate, the bit allocation range during encoding is controlled according to the amount of additional data other than audio data. Since the amount of audio data to be encoded is made variable and the additional data is inserted into the coded data stream, various data can be superimposed on the surplus bandwidth to make effective use of the bandwidth. The control of the bit allocation range executed by the bit allocation processing control stage 510 is performed on a frame basis, and the bit allocation range can be varied according to the data amount of the additional data buffer 512.

With these processes, even when additional data is inserted, the data insertion amount can be controlled in real time without deteriorating the sound quality in the pit allocation range.

(Embodiment 5) Next, a speech coding method and a speech coding apparatus according to Embodiment 5 of the present invention will be described with reference to the drawings. FIG. 16 is a block diagram showing a configuration of an encoder of a voice coding apparatus using a voice coding method according to the fifth embodiment. In the figure, the same reference numerals as those in FIG. 162 are independently operable encoding processing means A to C, 1
63 is a processing load value storage buffer for storing processing load value information of each of the encoding processing means A to C, and 164 is a sample data buffer for supplying sample data to each of the encoding processing means A to C. It is.

Next, the operation will be described. At the time of initialization before performing the encoding process, first, the sample data buffer 1
The predetermined sample data stored in H.64 is supplied to each of the encoding processing means A to C, and the processing load value of the encoding processing means A to C or the psychoacoustic model generated thereby is stored in the processing load value storage buffer 163. To be stored.

By outputting the processing load values in ascending or descending order, the C used in the device is output.
Quickly select an encoding processing means that matches the performance of the PU,
Encoding processing is performed by the encoding processing means. The contents of the encoding process are the same as those described in the first embodiment, and therefore will not be described here.

As described above, according to the present embodiment, at the time of initialization before the encoding process, each encoding processing means is operated using the sample data, the load value at that time is obtained, and the CPU to be used is obtained. Since the encoding processing means suitable for the processing capacity of the CPU is selected and used, the load on the CPU is reduced,
Optimum encoding processing can be performed.

In each of the above embodiments, the configuration realized by using a PC as an audio encoding device has been described as an example. However, for example, the audio encoding device may be incorporated in a device such as a VTR camera or a DVD encoder. The case can also be applied.

In each of the above embodiments, only audio is handled. However, when processing video together with audio, as shown in FIG. 17, in the configuration shown in FIG. A signal is input, and a video encoding unit 170 and an audio encoding unit 171 are provided instead of the low band encoding unit and the high band encoding unit.
This can be coped with by providing a system stream processing unit 172 in place of the bit stream forming processing unit. Using the configuration described above, based on control information from the outside, in the method described in each of the above embodiments, the operation amount of speech coding is changed, and a plurality of speech codes with different operation amounts are changed. By switching the scheme, it is possible to control the total amount of computation as the CPU. Alternatively, the processing amount of the video signal to be encoded may be changed according to the amount of the audio signal to be encoded.

Further, in addition to MPEG1 which performs subband coding, MPE which performs time / frequency conversion
G2, AAC, Dolby AC-3, ATRAC (M
Also in the case of performing a coding method such as D), as shown in FIG. 18, each means related to the encoding process is replaced with a first quantum information calculation means 181 and a second quantization information calculation means 182 having different calculation amounts. It is also possible to cope with the above by replacing these with the quantization means control means 180 and selecting and using these to handle the quantization information instead of the coded information.

[0089]

As described above, according to the audio coding system according to the first aspect of the present invention, the digital audio signal is divided into a plurality of frequency bands, and the audio coding system performs encoding for each band. Generating bit allocation information for each of the divided bands, each having a plurality of bit allocation means having different processing amounts, based on external control information, from among the plurality of bit allocation means,
Since the bit allocation means to be used is switched and the bit allocation is performed and the encoding is performed so that the processing is performed using the predetermined bit allocation means, the bit allocation means having the optimum processing amount is always selected. It is possible to perform encoding that does not exceed the processing amount of the CPU that can be occupied in the operating state, and the processing does not fall short of the input signal during real-time encoding. There is an effect that coding without interruption can be performed.

Further, according to the speech encoding method of the present invention, the load value representing the processing amount of the central processing unit which can be occupied for performing the encoding process is provided as the external control information. Using, based on the load value, with reference to a data table that stores in advance each processing amount when performing an encoding operation using each bit allocation means in the encoding process on the central processing unit, Since the bit allocation means is selected so as not to exceed the processing amount of the central processing unit that can be occupied by the encoding process, the central processing unit does not always receive a request that exceeds the operation capacity, and the entire system is not received. Has an effect that the control of can be performed smoothly.

According to a third aspect of the present invention, in the audio coding system according to the second aspect, the load value may be occupied for performing an encoding process. Since the processing amount control information from the monitoring means for monitoring the processing amount of the central processing unit is used, the bit allocation means having the optimum processing amount is selected within the range of the occupiable maximum performance of the central processing unit. Thus, there is an effect that the processing does not fall short of the input signal during the real-time encoding, that is, there is an effect that encoding can be performed without a break in the reproduced sound.

According to a fourth aspect of the present invention, in the audio coding system according to the first aspect of the present invention, the bit allocation processing by the bit allocation means is performed to improve the sound quality of the encoded data. Processing using a high-efficiency bit allocation method that performs bit allocation with high efficiency, and low-load bits that perform bit allocation with a low processing load with a small processing amount as compared with the processing using the high-efficiency bit allocation method. Since the process using the allocation method is performed, the encoder may appropriately switch the process of giving priority to the high sound quality of the encoded data or giving priority to the low load property of the encoding process over the sound quality. There is an effect that encoding that can be performed can be realized.

[0093] According to the speech encoding system of the fifth aspect of the present invention, in the speech encoding system of the first aspect, the switching of the bit allocating means used at the time of encoding is changed to an audio signal. Since the decoding is performed in frame units, which are the minimum units that can be decoded, even if the CPU occupancy of other applications sharing the CPU on the operating CPU suddenly increases during real-time encoding, the frame unit time can be reduced. This makes it possible to follow the processing amount of the CPU that can be occupied by the encoding process, and also to control the sound quality and the processing amount in real time.

[0094] According to the speech encoding system of the sixth aspect of the present invention, in the speech encoding system of the first aspect, each of the sub-band signals divided into a plurality of frequency bands is divided into a plurality of frequency bands. Grouping was performed so as to be a group consisting of a predetermined number of subband signals, and independent bit allocation processing was performed for each group, so that bit allocation information for each band was generated. There is an effect that encoding can be performed by selecting a bit allocation process according to the characteristics of each band.

According to the speech coding method of the present invention, in the speech coding method of the sixth aspect, the grouping is performed in the number of groups or in the frequency axis direction within the groups. The number of continuous subband signals is variably set to be the number specified by the external control information or the number specified based on the processing amount control information from the monitoring means. So C
There is an effect that the grouping can be dynamically performed according to the usage status of the PU.

[0096] According to the speech encoding system of the eighth aspect of the present invention, in the speech encoding system of the seventh aspect, the process of changing the number of subband signals can be decoded into an audio signal. Since it is performed in the frame unit which is the minimum unit, the change of the bit allocation method can be finely performed, and there is an effect that a highly accurate encoder can be realized.

According to the speech coding system of the ninth aspect of the present invention, in the speech coding system of the eighth aspect, at least one group in which bit allocation is not performed at the time of grouping is provided. Therefore, the number of groups, or the number of sub-band signals that are continuous in the frequency axis direction within the group, in a frame unit, which is the minimum unit that can be decoded into an audio signal, is the number specified by external control information, or By changing the number to the number specified by the processing amount control information from the monitoring unit, it is not necessary to perform the encoding process on the signal of the band belonging to the group to which bit allocation is not performed, and to belong to the group to which bit allocation is not performed. The bits to be allocated to a band can be distributed to other groups of bands where the bit allocation is made, Result, C encoding process occupies
There is an effect that the processing amount of the PU can be controlled and the sound quality of the band of another group to which bit allocation is performed can be improved.

[0098] According to the speech coding system of claim 10 of the present invention, in the speech coding system of claim 6, the subband signal is divided into groups.
For the sub-band signals grouped into the group belonging to the low band, the bit allocation is performed on the sub-band signals within the group using a high-efficiency bit allocation method capable of realizing high sound quality of the encoded data, For subband signals grouped into groups belonging to the band,
Using a low-load bit allocation process with a small processing amount compared to the high-efficiency bit allocation method, so as to perform bit allocation to sub-band signals in the group,
Higher sound quality of coded data can be achieved for low-bands that are sensitive to the human ear, while low-load bit allocation that prioritizes processing amount is assigned to high-bands that are insensitive to the human ear. As a result, it is possible to perform encoding with a reduced processing amount as a whole.

According to the speech coding system of the present invention, the number of bits that can be assigned to the bit assignment means independent for each group is determined in the speech coding system of the sixth embodiment. Allocatable bit calculation means is provided, and the number of bits that can be allocated to the entire group is determined for each group by using a ratio of each group to the entire group and weighting based on characteristics of each band belonging to each group. Since it is distributed to independent bit allocation means,
It is possible to perform encoding capable of performing bit allocation to bit allocation means of each group optimal for realizing high sound quality of encoded data in consideration of auditory characteristics with respect to characteristics of an input signal or each band. effective.

According to the speech coding system of the present invention, weighting based on the characteristics of each band belonging to each group is performed in each speech coding system. The weighting based on the predetermined minimum audible limit value for each has an effect that a meaningful and effective bit allocation process can be performed when the human listens.

According to the speech coding system of the present invention, the weighting based on the characteristics of each band belonging to each group is performed by the input digital coding system. The weighting based on the subband signal level of each frequency band belonging to each group obtained by performing subband analysis on the audio signal has an effect that an effective bit allocation process can be performed.

According to a fourteenth aspect of the present invention, in the audio coding system according to the eleventh aspect, the weighting based on the characteristics of each band belonging to each group is applied to the input digital audio signal. By performing weighting based on the spectrum signal level belonging to each group, which is obtained by performing a linear conversion on the group, there is an effect that an effective bit allocation process can be performed.

According to a speech coding method according to claim 15 of the present invention, in the speech coding method according to claim 6, the signal level belonging to each group is set to a high level equal to or higher than a predetermined threshold value. Bit allocation using a highly efficient bit allocation method capable of realizing high-quality encoded data for a low-level signal, and the signal level belonging to each group is a low-level signal of a predetermined threshold or less. In contrast, the bit allocation is performed using a low-load bit allocation method that requires less processing compared to the above-described high-efficiency bit allocation method, so that a signal that is not so important as encoded data compared to other bands is used. On the other hand, there is an effect that it is possible to perform encoding that can achieve high sound quality of encoded data without dividing a processing load.

According to a speech coding system of the present invention, in the speech coding system of the present invention, a signal level belonging to each group is converted into an input digital audio signal by sub-band analysis. By applying the sub-band signal level obtained by performing the above, there is an effect that an effective bit allocation process can be performed.

According to the speech coding system of claim 17 of the present invention, in the speech coding system of claim 15, the signal level belonging to each group is converted linearly into an input digital audio signal. Since the spectrum signal level is obtained by performing the above, there is an effect that it is possible to perform an effective bit allocation process.

According to the speech coding system of the present invention, in the speech coding system of the present invention, the signal level belonging to each of the groups is set to a minimum audible level for each predetermined band. Since the threshold value is set, there is an effect that the effective and effective bit allocation processing can be performed when the human listens.

According to the speech coding method of claim 19 of the present invention, in the speech coding method of any one of claims 4, 10, and 15, it is possible to improve the sound quality of the encoded data. As feasible high-efficiency bit allocation processing, bit allocation is performed using a relationship between a signal and a mask ratio value based on a predetermined psychoacoustic model. As a low-load bit allocation process with a small number of bits, bit allocation is performed by adding a predetermined minimum audible limit value for each band to a signal level divided into a plurality of frequency bands. As long as the sound quality is not impaired, the processing amount of the system can be reduced.

According to a twentieth aspect of the present invention, there is provided a speech encoding system according to the nineteenth aspect, wherein the psychoacoustic model is a psychoacoustic model specified by MPEG. (Motion
The same effect as described above can be obtained even in audio encoding processing using Picture Experts Group.

According to a twenty-first aspect of the present invention, in the audio coding system according to the fifth or eighth aspect, a frame which is a minimum unit which can be decoded into the audio signal is an MPEG signal. (Motion Picture E
xperts Group), the same effect as described above can be obtained even in audio encoding processing using MPEG.

[0110] According to a speech coding method according to claim 22 of the present invention, in the speech coding method according to claim 1, the bit allocation means assigns a predetermined value to each of the divided bands. Generates bit allocation information based on information output from the psychoacoustic model,
Once every N (N = 1, 2, 3,...) Frames, bit allocation information is generated based on information output from the predetermined psychoacoustic model. On the other hand, bit allocation information is generated and encoded based on the information output from the psychoacoustic model and the signal information of each of the divided bands, so that the CPU load in the time axis direction is reduced. There is an effect that it can be reduced.

[0111] According to the speech coding method of claim 23 of the present invention, the speech coding method of claim 1 has an auditory psychological model whose processing amount can be controlled step by step. Based on external control information, the processing amount of the psychoacoustic model is controlled, and bit allocation information for each band is generated so that processing is performed using the psychoacoustic model of a predetermined processing amount. Therefore, there is an effect that CPU load control in consideration of an auditory effect can be performed.

Further, according to the speech coding method of claim 24 of the present invention, the speech coding method of claim 1 has a plurality of psychoacoustic models each having a different processing amount, and is controlled by an external control. Based on the information, a psychoacoustic model to be used is switched to generate bit allocation information for each band so that processing is performed using a predetermined psychoacoustic model from among the plurality of psychoacoustic models. , So it is easier to add auditory effects to C
There is an effect that PU load control can be performed.

Further, according to the speech coding system of the present invention, a digital audio signal is divided into a plurality of frequency bands, bit allocation information for each of the divided bands is generated, and a predetermined bit rate is determined. This is a voice coding method that performs coding for each band for the purpose of transmission in the CDMA system, and controls the bit allocation range in a frame where data is inserted into the coded data stream, and variably controls the amount of coded audio data C
The effect is obtained that the load of the PU processing becomes constant, and that various data can be superimposed on the surplus bandwidth to effectively use the bandwidth.

According to the speech coding system of the present invention, in the speech coding system of the present invention, the bit allocation range is controlled on a frame basis based on external control information. Since the amount of encoded audio data is variably controlled, the processing load on the CPU can be reduced efficiently.

According to the speech coding system of claim 27 of the present invention, in the speech coding system of claim 26, monitoring means for monitoring a buffer of additional data as the external control information. Is used, the load data can be preferentially superimposed.

[0116] According to the speech encoding system of claim 28 of the present invention, in the speech encoding system of claim 1, depending on the performance of the central processing unit in which the encoding process is executed, Since the processing load value information of the plurality of bit allocation means or the plurality of psychoacoustic models is output to the outside at the time of initialization before the encoding processing operation, the CPU used before the actual encoding is performed. Information about the performance of the CPU can be obtained, and the processing load on the CPU can be efficiently reduced.

According to the speech coding system of claim 29 of the present invention, in the speech coding system of claim 28, a plurality of bit allocation means output as information to the outside, or a plurality of Since each piece of processing load value information of the psychoacoustic model is output in descending order or ascending order, there is an effect that the encoding processing means can be selected quickly.

Further, according to the audio coding method according to claim 30 of the present invention, the coding processing of the video signal and the audio signal is different in the audio coding method in which the processing is performed by the same central processing unit. The encoding is performed with a plurality of computation amounts, and the overall computation amount of processing on the central processing unit is controlled by changing the encoding amount of the audio signal or the video signal. As a result, there is an effect that the load processing of the CPU can be performed also in the process of encoding the video signal together with the audio.

Further, according to the audio coding method according to claim 31 of the present invention, in the audio coding method in which the encoding processing of the video signal and the audio signal is performed by the same central processing unit, The encoding is performed using a plurality of encoding schemes having different amounts, and by changing the encoding scheme for encoding the audio signal, the overall operation of the processing on the central processing unit is performed. Because I controlled the amount,
In the process of encoding a video signal together with audio, C
There is an effect that the load processing of the PU can be performed.

According to the speech coding system of claim 32 of the present invention, in the speech coding system of claim 30 or 31, control of processing on the central processing unit is externally controlled. Is performed based on the control information described above, so that the processing load on the CPU can be efficiently reduced.

Further, according to the speech coding method of claim 33 of the present invention, the speech coding method performs time / frequency conversion on a digital audio signal and generates quantization information to perform coding. A plurality of quantization information calculation means each having a different operation amount, and a predetermined quantization information calculation means is selected from the plurality of quantization information calculation means based on control information from the outside. Since the quantization information calculation means to be used is switched so as to select and perform the processing, the quantization information is calculated and the coding is performed. Therefore, in the coding processing apparatus for performing the coding of the time / frequency conversion method, This also has the effect that the processing load on the CPU can be reduced.

According to a speech coding apparatus according to claim 34 of the present invention, speech coding is performed using the speech coding method according to any one of claims 1 to 33. Therefore, the VT incorporating the audio coding method
There is an effect that the same effect as described above can be obtained in a device such as an R camera.

Also, in the data storage medium according to claim 35 of the present invention, since the steps of the audio coding method according to any one of claims 1 to 33 are recorded, the data storage medium is used. By incorporating the above-mentioned speech coding system into the apparatus, the same effect as described above can be obtained.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration of an entire system using a personal computer in order to realize a speech encoding device using a speech encoding method according to a first embodiment of the present invention.

FIG. 2 is a block diagram illustrating a configuration of an encoder included in the speech encoding device according to the first embodiment.

FIG. 3 is a block diagram showing a detailed configuration of high-band coding processing means constituting the encoder.

FIG. 4 is a block diagram illustrating a more detailed configuration of an encoder included in the speech encoding device according to the first embodiment.

FIG. 5 is a schematic diagram showing an example of a bit allocation process for each group used in the audio coding scheme according to the first embodiment.

FIG. 6 is a schematic diagram showing another example of a bit allocation process for each group used in the audio coding method according to the first embodiment.

FIG. 7 is a diagram showing a flow for describing an encoding operation of an encoder included in the speech encoding device according to the first embodiment.

FIG. 8 is a schematic diagram showing an example in which a bit allocation process for each group is performed using a threshold value, which is used in the audio encoding method according to the first embodiment.

FIG. 9 is a block diagram illustrating a detailed configuration of a modified example of the encoder included in the speech encoding device according to the first embodiment of the present invention.

FIG. 10 is a block diagram illustrating a data storage medium according to a second embodiment of the present invention and a configuration in a case where a speech encoding device is configured using the storage medium.

FIG. 11 is a block diagram illustrating a configuration of an encoder that forms a conventional speech encoding device.

FIG. 12 is a diagram illustrating a detailed configuration of a low-band coding processing unit included in a speech coding device according to a third embodiment of the present invention.

FIG. 13 is a diagram for explaining the state of the psychoacoustic model in each frame at the time of low-band coding by the speech coding apparatus according to the third embodiment.

FIG. 14 is a diagram illustrating a detailed configuration of a low-band coding processing unit included in a speech coding device according to a fourth embodiment of the present invention.

FIG. 15 is a diagram illustrating an example of a bit allocation process using the speech encoding device according to the fourth embodiment.

FIG. 16 is a block diagram illustrating a configuration of an encoder included in a speech encoding device according to a fifth embodiment of the present invention.

FIG. 17 is a block diagram illustrating a configuration of an encoder when a video signal is handled together with an audio signal.

FIG. 18 is a block diagram illustrating a configuration in a case where the present invention is applied to encoding processing in an encoding processing device that performs time / frequency conversion coding.

[Explanation of symbols]

 DESCRIPTION OF SYMBOLS 1 Personal computer (sound encoding device) 11 HDD 12a PDD 12b FDD 13 Memory 14 CPU (Central processing unit) 15 Data bus 16 Video capture card 17 Camera 18 Sound card 19 Microphone 20 Encoder 21 CPU load monitoring information 22 Code Coding means control means 23 low-band coding processing means 24 high-band coding processing means 25 bit stream forming processing means 26 coding mode designation signal 101 coder 102 sub-band analysis means 103 scale factor extraction means 104 FFT means 105 psychoacoustic Analysis means 106 quantization / encoding means 107 auxiliary information encoding means 108 bit stream forming means 109 band output adaptive bit allocation means 110 psychoacoustic model bit allocation means 111 group Processing means 112 bit allocation processing control means 121 processing amount control information 160 to 162 coding processing means A to C 163 processing load value storage buffer 164 sample data buffer 170 video coding processing means 171 audio coding processing means 172 system stream forming processing Means 180 Quantization means control means 181 First quantization information calculation means 182 Second quantization information calculation means FC Floppy disk case FD Floppy disk D Floppy disk body Se Sector Tr Track Cs Computer system FDD Floppy disk drive

Claims (35)

[Claims] 1. An audio coding method for dividing a digital audio signal into a plurality of frequency bands and performing encoding for each band, wherein bit allocation information for each of the divided bands is generated, and A plurality of different bit allocation means, and based on control information from the outside, a bit allocation means to be used is selected from the plurality of bit allocation means so that processing is performed using a predetermined bit allocation means. A speech coding method characterized by performing coding by switching and performing bit allocation. 2. The speech coding method according to claim 1, wherein a load value representing a processing amount of a central processing unit that can be occupied for performing an encoding process is used as the external control information. Based on the above, the encoding process is occupied by referring to a data table that stores in advance the processing amount when the encoding operation is performed using each bit allocation unit in the encoding process on the central processing unit. A speech coding method, wherein the bit allocation means is selected so as not to exceed a processing amount of a central processing unit which can be processed. 3. The speech coding method according to claim 2, wherein the load value is a processing amount control information from a monitoring unit that monitors a processing amount of the central processing unit that can be occupied for performing an encoding process. A speech coding method characterized by using. 4. The high-efficiency bit allocation method according to claim 1, wherein the bit allocation processing by the bit allocation means performs high-efficiency bit allocation capable of realizing high-quality sound of encoded data. And a process using a low-load bit allocation method for performing bit allocation with a low load with a smaller processing amount compared to the process using the high-efficiency bit allocation method. System. 5. The audio encoding system according to claim 1, wherein the switching of the bit allocation means used at the time of encoding is performed in a frame unit which is a minimum unit that can be decoded into an audio signal. Encoding method. 6. A speech coding system according to claim 1, wherein the sub-band signals of each band divided into a plurality of frequency bands are formed into groups each including a predetermined number of sub-band signals. A speech coding method characterized by performing grouping, performing independent bit allocation processing for each group, and generating bit allocation information for each band. 7. The audio coding method according to claim 6, wherein the grouping is performed by specifying the number of groups or the number of subband signals continuous in the frequency axis direction within the group by the external control information. A voice coding method, which is variably performed such that the number becomes the specified number or the number specified based on the processing amount control information from the monitoring means. 8. The audio encoding method according to claim 7, wherein the process of changing the number of subband signals is performed in frame units, which are minimum units that can be decoded into audio signals. . 9. The speech encoding method according to claim 8, wherein at least one group to which bit allocation is not performed is provided at the time of grouping. 10. The audio coding method according to claim 6, wherein the sub-band signals are grouped to realize high-quality encoded data for the sub-band signals grouped into a group belonging to a low band. By using a process for performing a possible high-efficiency bit allocation, the bits are allocated to the sub-band signals in the group. A voice coding method characterized by performing bit allocation to subband signals in a group using low-load bit allocation processing with a relatively small processing amount. 11. A speech coding method according to claim 6, further comprising: assignable bit operation means for determining the number of assignable bits to the independent bit assignment means for each group, wherein the ratio of each group to the entire group is: Speech coding characterized by distributing the number of bits that can be allocated to the entire group to independent bit allocating means for each group using weights based on characteristics of each band belonging to each group. method. 12. The speech coding method according to claim 11, wherein the weighting based on the characteristics of each band belonging to each group is weighted based on a predetermined minimum audible limit value for each band. Audio coding method. 13. A speech coding method according to claim 11, wherein weights based on characteristics of each band belonging to each group are obtained by subjecting an input digital audio signal to sub-band analysis. A speech coding method characterized by weighting based on a subband signal level of a frequency band. 14. The speech coding method according to claim 11, wherein weighting based on characteristics of each band belonging to each group is performed by performing a linear conversion on an input digital audio signal, and the spectrum signal belonging to each group is obtained. A speech coding method characterized by weighting based on levels. 15. The audio coding method according to claim 6, wherein for a high-level signal whose signal level belonging to each group is equal to or more than a predetermined threshold value, it is possible to realize high-quality encoded data. Bit allocation is performed using a high-efficiency bit allocation method, and a signal level belonging to each group whose signal level is lower than or equal to a predetermined threshold is processed in a smaller amount than the high-efficiency bit allocation method. A voice coding method characterized in that bit allocation is performed by using a low-load bit allocation method with less noise. 16. The audio coding method according to claim 15, wherein the signal level belonging to each group is a subband signal level obtained by performing a subband analysis on an input digital audio signal. Encoding method. 17. The speech coding method according to claim 15, wherein the signal level belonging to each group is a spectrum signal level obtained by performing a linear conversion on an input digital audio signal. method. 18. The speech coding system according to claim 15, wherein the signal levels belonging to each group are set to a minimum audible limit value for each predetermined band. 19. The speech encoding method according to claim 4, wherein the high-efficiency bit allocation process capable of realizing high-quality sound of the encoded data is based on a predetermined psychoacoustic model. A bit allocation process performed using a relationship with a signal-to-mask ratio value. The low-load bit allocation process with a smaller processing amount compared to the high-efficiency bit allocation process is performed on a signal divided into a plurality of frequency bands. A speech coding method, which is a bit allocation process performed by adding a predetermined minimum audible limit value for each band to a level. 20. The audio coding method according to claim 19, wherein the psychoacoustic model is MPEG (Motion Picture Exposure).
erts Group). 21. The audio coding method according to claim 5, wherein a frame that is a minimum unit that can be decoded into the audio signal is a frame specified by an MPEG (Motion Picture Experts Group). Characteristic speech coding method. 22. The speech coding method according to claim 1, wherein the bit allocation means generates bit allocation information for each of the divided bands based on information output from a predetermined psychoacoustic model. Generating bit allocation information once every N (N = 1, 2, 3,...) Frames based on information output from the predetermined psychoacoustic model, and generating the bit allocation information. For a frame that did not exist, bit allocation information is generated based on the information output from the psychoacoustic model and the signal information of each of the divided bands, and coding is performed. . 23. The speech encoding method according to claim 1, further comprising: an psychoacoustic model capable of controlling a processing amount in a stepwise manner, and a processing amount control of the psychoacoustic model based on external control information. And generating bit allocation information for each band such that processing is performed using a psychoacoustic model of a predetermined processing amount. 24. The speech encoding method according to claim 1, comprising a plurality of psychoacoustic models each having a different processing amount, and based on control information from the outside, a predetermined psychoacoustic model is selected from the plurality of psychoacoustic models. Switch the psychoacoustic model to be used so that processing is performed using the psychoacoustic model,
A speech coding method for generating bit allocation information for each band. 25. A speech code which divides a digital audio signal into a plurality of frequency bands, generates bit allocation information for each of the divided bands, and encodes each band for transmission at a predetermined bit rate. An audio encoding method, comprising: controlling a bit allocation range in a frame into which data is inserted into an encoded data stream; and variably controlling an encoded audio data amount. 26. The audio encoding method according to claim 25, wherein a bit allocation range is controlled in frame units based on external control information, and an encoded audio data amount is variably controlled. Audio coding method. 27. A speech coding method according to claim 26, wherein data control information from monitoring means for monitoring a buffer for additional data is used as said external control information. . 28. The audio coding method according to claim 1, wherein a plurality of bit allocation means or a plurality of bit allocation means are provided at the time of initialization before the encoding processing operation, depending on the performance of the central processing unit on which the encoding processing is performed. A speech encoding method for outputting each processing load value information of the psychoacoustic model of the present invention to the outside. 29. The speech coding method according to claim 28, wherein a plurality of bit allocation means output as information to the outside or processing load value information of a plurality of psychoacoustic models are output in descending order or ascending order. A speech coding method characterized by being performed. 30. An audio encoding method in which encoding processing of a video signal and an audio signal is performed by the same central processing unit, wherein encoding is performed with a plurality of different computational amounts. An audio coding method characterized by controlling the overall calculation amount of processing on the central processing unit by changing the calculation amount of signal or video signal coding. 31. An audio encoding system in which video and audio signals are encoded by the same central processing unit, wherein the encoding is performed by using a plurality of encoding systems having different arithmetic operations. By changing the encoding method of the audio signal encoding,
A speech coding system for controlling an overall operation amount of processing on the central processing unit. 32. The speech coding method according to claim 30, wherein control of processing on the central processing unit is performed based on control information from outside. 33. For a digital audio signal,
A speech coding method for performing coding by performing time / frequency conversion and generating quantization information. The coding method includes a plurality of quantization information calculation units each having a different operation amount, and is based on control information from outside. The quantization information calculation means to be used is switched to calculate the quantization information by switching the quantization information calculation means to be used so that a predetermined quantization information calculation means is selected from the plurality of quantization information calculation means and the processing is performed. A speech coding method characterized by performing coding. 34. A speech encoding apparatus that performs speech encoding using the speech encoding method according to claim 1. Description: 35. A data recording medium on which is recorded the steps of the speech encoding method according to claim 1. Description: